Multiple access points (APs) are much more likely to be available for stations (STAs) due to the popularity of wireless LANs. The serious issue of how an appropriate AP is selected from those that are available in a wireless LAN therefore arises. We discuss the development of a decentralized architecture for selecting APs, and examine its fundamental characteristics. The proposed architecture should be introduced without adversely affecting the performance of the existing common architecture that is currently being deployed. Therefore, the deployability of our architecture is examined in this respect. Furthermore, the dynamic behavior of the proposed architecture is studied in addition to static characteristics to evaluate its robustness against various dynamic changes in situation due to AP breakdowns and bursty arrivals of STAs. Simulations revealed that the proposed architecture can attain excellent performance in all the cases treated here.

The performance of a real-time networked application can be drastically affected by delays in packets traversing the network. Some real-time applications impose limits for acceptable network delay, and so a packet which is delayed longer than the limit before arriving at its destination is worthless to the flow to which the packet belongs. Not only that, but the rejected packet is also damaging to the quality of other flows in the network, because it may increase the queuing delay for other packets. Therefore, this paper proposes an adaptive scheme using two mechanisms, in which packets experiencing too great a delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. This earlier discarding of packets is expected to improve the overall delay performance of real-time flows competing for network resources when the network is congested. An extensive simulation is conducted, and the results show that the scheme has great potential in improving the delay performance of real-time traffic in both homogeneous and heterogeneous environments in terms of traffic volume and application delay requirements.

As the Internet role changes from the experimental environment to the social infrastructure, end-to-end quality for data transfer of various types of traffic has been required as well as its connectivity. So we should precisely measure some performance such as packet loss probability and delay at routers on some source-destination path. By using so-called passive measurement technique, we can get the queue length distribution from some routers individually and estimate the end-to-end transmission delay. However, there may be some correlation between queue lengths of two or more routers packets go through in sequence, which would lead to inaccurate estimation of end-to-end delay performance. Thus in this paper, we model two tandem routers as queueing system, and analyze the queue length distributions and their correlation. Through some numerical results, we will investigate the impact of traffic parameters on the degree of correlation and how it affects the estimation of delay performance.

A transport layer mobility management scheme for handling seamless handoffs between appropriate networks is presented. The future mobile environment will be characterized by multimodal connectivity with dynamic switching. Many technologies have been proposed to support host mobility across diverse wireless networks, and operate in various layers of the network architecture. Our major focus is on the transport protocol that recovers packets lost during handoffs and controls transmission speed to achieve efficient communication. Majority of the existing technologies can maintain the connection by updating the information of a single connection around a handoff. Moreover, none of the studies extensively examine the handoff latencies and focus how an appropriate network is selected, during the handoff. In this paper, we first extensively investigate the various handoff latencies and discuss the limited performance of existing technologies based on the single connection. We then propose a new scheme resolving the problems by the transport protocol enabling the adaptive selection of an appropriate interface based on communication condition among all available interfaces. Finally, we demonstrate that the proposed scheme promptly and reliably selects the appropriate interface, and achieves excellent goodput performance by comparing with the existing technologies.

We present the concept of folksonomical peer-to-peer (P2P) file sharing networks that allow participants (peers) to freely assign structured search tags to files. These networks are similar to folksonomies in the present Web from the point of view that users assign search tags to information distributed over a network. As a concrete example, we consider an unstructured P2P network using vectorized Kansei (human sensitivity) information as structured search tags for file search. Vectorized Kansei information as search tags indicates what participants feel about their files and is assigned by the participant to each of their files. A search query also has the same form of search tags and indicates what participants want to feel about files that they will eventually obtain. A method that enables file search using vectorized Kansei information is the Kansei query-forwarding method, which probabilistically propagates a search query to peers that are likely to hold more files having search tags that are similar to the query. The similarity between the search query and the search tags is measured in terms of their dot product. The simulation experiments examine if the Kansei query-forwarding method can provide equal search performance for all peers in a network in which only the Kansei information and the tendency with respect to file collection are different among all of the peers. The simulation results show that the Kansei query forwarding method and a random-walk-based query forwarding method, for comparison, work effectively in different situations and are complementary. Furthermore, the Kansei query forwarding method is shown, through simulations, to be superior to or equal to the random-walk based one in terms of search speed.

With the approval of IEEE 1901 standard for power line communications (PLC) and the recent Internet-enable home appliances like the IPTV having access to a content-on-demand service through the Internet as AcTVila in Japan, there is no doubt that PLC has taken a great step forward to emerge as the preeminent in-home-network technology. However, existing schemes developed so far have not considered the PLC network connected to an unstable Internet environment (i.e. more realistic situation). In this paper, we investigate the communication performance from the end-user's perspective in networks with large and variable round-trip time (RTT) and with the existence of cross-traffic. Then, we address the problem of unfair bandwidth allocation when multiple and different types of flows coexist and propose a TCP rate control considering the difference in terms of end-to-end delay to solve it. We validate our methodology through simulations, and show that it effectively deals with the throughput unfairness problem under critical communication environment, where multiple flows with different RTTs share the PLC and cross-traffic exists on the path of the Internet.

On wide area networks (WANs), UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission. Meanwhile, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol (TCP) traffic. As opposed to UDP's simplicity, TCP adopts a unique flow control mechanism with sliding windows. Hence, the quality of service (QoS) of real-time applications using UDP is affected by TCP traffic and its flow control mechanism whenever TCP and UDP share a bottleneck node. In this paper, the characteristics of UDP packet loss are investigated through simulations of WANs conveying UDP and TCP traffic simultaneously. In particular, the effects of TCP flow control on the packet loss of real-time audio are examined to discover how real-time audio should be transmitted with the minimum packet loss, while it is competing with TCP traffic for the bandwidth. The result obtained was that UDP packet loss occurs more often and successively when the congestion windows of TCP connections are synchronized. Especially in this case, the best performance of real-time audio applications can be obtained when they send-small sized packets without reducing their transmission rates.

Since congestion is very likely to happen in the Internet, locating congested areas (path segments) along a congested path is vital to appropriate actions by Internet Service Providers to mitigate or prevent network performance degradation. We propose a practical method to locate congested segments by actively measuring one-way end-to-end packet losses on appropriate paths from multiple origins to multiple destinations, using a network tomographic approach. Then we conduct a long-term experiment measuring packet losses on multiple paths over the Japanese commercial Internet. The experimental results indicate that the proposed method is able to precisely locate congested segments. Some findings on congestion over the Japan Internet are also given based on the experiment.

In ubiquitous networks, Mobile Nodes (MNs) often suffer from performance degradation due to the following two reasons: (1) reduction of signal strength by the movement of an MN and intervening objects, and (2) radio interference with other WLANs. Therefore, handover initiation based on quick and reliable detection of the deterioration in a wireless link condition arising from the above two reasons is essential for achieving seamless handover. In previous studies, we focused on a handover decision criterion and described the problems related to the two existing decision criteria. Furthermore, we showed the effectiveness of the number of frame retransmissions through simulation experiments. However, a comparison of the signal strength and the number of frame retransmissions could not be examined due to the unreliability of the signal strength in simulations. Therefore, in the present paper, by employing FTP and VoIP applications, we compare the signal strength and the number of frame retransmissions as a handover decision criterion with experiments in terms of (1) and (2) in a real environment. Finally, we clarify the problem of the signal strength in contrast to the effectiveness of the number of frame retransmissions as a handover decision criterion.

Traffic Engineering (TE) is important for improving QoS in forwarding paths by efficient use of network resources. In fact, MPLS allows several detour paths to be (pre-)established for some source-destination pair as well as its primary path of minimum hops. Thus, we focus on a two-phase path management scheme using these two kinds of paths. In the first phase, each primary path is allocated to a flow on a specific source-destination pair if the path is not congested, i.e., if its utilization is less than some predetermined threshold; otherwise, as the second phase, one of the detour paths is allocated randomly if the path is available. Therefore, in this paper, we analytically evaluate this path management scheme by extending the M/M/c/c queueing system, and through some numerical results we investigate the impact of a threshold on the flow-blocking probability. Through some numerical results, we discuss the adequacy of the path management scheme for MPLS-TE.

There is an emerging requirement for real-time flow-based traffic monitoring, which is vital to detecting and/or tracing DoS attacks as well as troubleshooting and traffic engineering in the ISP networks. We propose the architecture for a scalable real-time flow measurement tool in order to allow operators to flexibly define "the targeted flows" on-demand, to obtain various statistics on those flows, and to visualize them in a real-time manner. A traffic distribution device and multiple traffic capture devices processing packets in parallel are included in the architecture, in which the former device copies traffic and distributes it to the latter devices. We evaluate the performance of a proto-type implementation on PC-UNIX in testbed experiments to demonstrate the scalability of our architecture. The evaluation shows that the performance increases in proportion to the number of the capture devices and the maximum performance reaches 80 K pps with six capture devices. Finally we also show applications of our tool, which indicate the advantage of flexible fine-grained flow measurements.

In this paper, we propose a file replication method to achieve load balancing in terms of write access to storage device ("write storage access load balancing" for short) in unstructured peer-to-peer (P2P) file-sharing networks in which the popularity trend of queried files varies dynamically. The proposed method uses a write storage access ratio as a load balance index value in order to stabilize dynamic P2P file-sharing environments adaptively. In the proposed method, each peer autonomously controls the file replication ratio, which is defined as a probability to create the replica of the file in order to uniform write storage access loads in the similar way to thermal diffusion phenomena. Theoretical analysis results show that the behavior of the proposed method actually has an analogy to a thermal diffusion equation. In addition, simulation results reveal that the proposed method has an ability to realize write storage access load balancing in the dynamic P2P file-sharing environments.

In the present paper, we propose an evolutionary P2P networking technique that dynamically and adaptively optimizes several P2P network topologies, in which all of the nodes are included at the same time, in an evolutionary manner according to given evaluation criteria. In addition, through simulations, we examine whether the proposed evolutionary P2P networking technique can provide reliable search capability in dynamic P2P environments. In simulations, we assume dynamic P2P environments in which each node leaves and joins the network with its own probability and in which search objects vary with time. The simulation results show that topology reconstruction by the evolutionary P2P networking technique is better than random topology reconstruction when only a few types of search objects are present in the network at any moment and these search objects are not replicated. Moreover, for the scenario in which the evolutionary P2P networking technique is more effective, we show through simulations that when each node makes several links with other nodes in a single network topology, the evolutionary P2P networking technique improves the reliable search capability. Finally, the number of links that yields more reliable search capability appears to depend on how often nodes leave and join the network.

In ubiquitous networks based on Wireless Local Area Networks (WLANs) with limited individual coverage, mobile nodes will be likely to traverse different WLANs during TCP communication. An effective handover management scheme for achieving seamless and efficient communication throughout the handover operation is therefore crucial. To achieve this, the following three requirements are essential: (i) early initiation of handover, (ii) elimination of communication interruption upon handover, (iii) selection of an optimal WLAN. The handover scheme proposed in this study employs frame retransmission over WLAN as an indicator of link degradation, and a handover manager (HM) on the transport layer obtains the number of frame retransmissions on the MAC layer using a cross-layer architecture in order to achieve (i) and (iii). Then, it also employs multi-homing in order to achieve (ii). Simulations demonstrate that the proposed scheme can satisfy all of the three requirements and is capable of maintaining TCP performance throughout the handover operation.

In the near future, wireless local area networks (WLANs) will overlap to provide continuous coverage over a wide area. In such ubiquitous WLANs, a mobile node (MN) moving freely between multiple access points (APs) requires not only permanent access to the Internet but also continuous communication quality during handover. In order to satisfy these requirements, an MN needs to (1) select an AP with better performance and (2) execute a handover seamlessly. To satisfy requirement (2), we proposed a seamless handover method in a previous study. Moreover, in order to achieve (1), the Received Signal Strength Indicator (RSSI) is usually employed to measure wireless link quality in a WLAN system. However, in a real environment, especially if APs are densely situated, it is difficult to always select an AP with better performance based on only the RSSI. This is because the RSSI alone cannot detect the degradation of communication quality due to radio interference. Moreover, it is important that AP selection is completed only on an MN, because we can assume that, in ubiquitous WLANs, various organizations or operators will manage APs. Hence, we cannot modify the APs for AP selection. To overcome these difficulties, in the present paper, we propose and implement a proactive AP selection method considering wireless link condition based on the number of frame retransmissions in addition to the RSSI. In the evaluation, we show that the proposed AP selection method can appropriately select an AP with good wireless link quality, i.e., high RSSI and low radio interference.

We examine delay performance of packets from constant bit rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the Σ Di/G/1 queue with vacations. Our analysis allows heterogeneous service time and heterogeneous interarrival time. Thus, we can get the impact of packet length of a stream on the delay time of other streams. We then give various numerical results for enterprise multimedia networks, which include voice, video and data communication services. From our quantitative evaluation, we conclude that packet length of video traffic has large influence on the delay time of voice traffic while voice traffic gives a little impact on the delay time of video traffic.

Over the future Internet, the real time communication generating such as CBR (Constant Bit Rate) traffic will widely spread, whereas the current Internet has no ability to provide QoS (Quality of Service) assurance for real time communication so far. In QoS networks, CBR traffic will have priority for its stringent QoS requirement over non-real time traffic such as TCP connections, which use the unused bandwidth left by CBR connections. Therefore, there is possibility that CBR traffic with priority causes TCP throughput degradation in QoS networks. For this reason, the performance of Tahoe TCP has been examined in that context, but other TCP variants such as Reno TCP, NewReno TCP and TCP with SACK option, which are now very common, have not yet been investigated clearly. In the present research, we will clarify how these TCP variants behave in QoS networks by means of simulations and compare their performance. From the results, SACK TCP can adapt very well to the changing bandwidth available and is very robust against the fluctuation, i.e., burstness, of CBR packet arrival process.

Most of the QoS-based routing schemes proposed so far focus on improving the performance of individual service classes. In a multi-class network where high priority QoS traffic coexists with best-effort traffic, routing decision for QoS sessions will have an effect on lower ones. A mechanism that allows dynamic sharing of link resources among multiple classes of traffic is needed. In this paper we propose a multi-class routing algorithm based on inter-class sharing resources among multiple class of traffic. Our algorithm is based on the concept of "the virtual residual bandwidth," which is derived from the real residual bandwidth. The virtual residual bandwidth is greater than the residual bandwidth when the load of lower priority traffic is light, and smaller when the load of lower priority traffic is heavy. The idea of our approach is simple since the routing algorithm for individual traffic doesn't change, but the only change is the definition of the link cost. We demonstrate through some extensive simulations the effectiveness of our approach when the best effort distribution is uneven and when its load is heavy. Also better performance is noticed when using topology with large number of alternative paths.